Re: [asterisk-users] FXS ports on TDM410P card...
On Wednesday 12 October 2011, Michael C. Robinson wrote: > My analog card, uses a PCI slot and a 12V power connector, is configured > with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS > ports but I can't dial out from them. Is extensions.conf where I need > to make changes? If you can't make calls *from* a phone, but you can make calls *to* it, that suggests a problem with its default context. Your configuration snippets shew "myphones" as the default context in chan_dahdi.conf, but the context in the dialplan was "my-phones". Make them match up, reload all configuration files and it should all Just Work. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] failed to extend from 512 to 676
Hi all. I've got Asterisk 1.6.2.9 with RT SIP running, and for the most part, it's working great. However, once in a while, I'll get some strange output from "sip show peer .." For example: === *CLI> sip show peer 687F74D9848C-1 * Name : 687F74D9848C-1 Realtime peer: Yes, cached Secret : MD5Secret: Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : [Oct 11 18:00:48] failed to extend from 512 to 676 [Oct 11 18:00:48] failed to extend from 512 to 676 [Oct 11 18:00:48] failed to extend from 512 to 676 [Oct 11 18:00:48] failed to extend from 512 to 676 [Oct 11 18:00:48] failed to extend from 512 to 676 [Oct 11 18:00:48] failed to extend from 512 to 676 [Oct 11 18:00:48] failed to extend from 512 to 676 === I'm not able to figure out what causes this and usually, I have to prune the client, or reload sip to make it stop. Can someone tell me what causes it and how to fix it? TIA, -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to asterisk IAX trunk
Thanks for your quick replies guys, I'll give your suggestions a try
On Tue, 2011-10-11 at 15:02 +0100, Roger Burton West wrote:
> On Tue, Oct 11, 2011 at 02:53:26PM +0100, Jonathan Archer wrote:
>
> >How can I get the 5 to stay where it is so that lookups work correctly?
> >is it part of the outbound CID?
>
> My trunking (prefix 9 to get trunk access from either side of the link)
> includes things like:
>
> exten => _9NX.,1,Set(CALLERID(num)=9${CALLERID(num)})
> exten => _9NX.,n,Dial(IAX2/remoteserver/${EXTEN:1},,wW)
>
> R
>
> --
> _
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Re: [asterisk-users] failed to extend from 512 to 676
> I've got Asterisk 1.6.2.9 with RT SIP running, and for the most part, > it's working great. > > However, once in a while, I'll get some strange output from "sip show > peer .." > > For example: > === > *CLI> sip show peer 687F74D9848C-1 > * Name : 687F74D9848C-1 > Realtime peer: Yes, cached > Secret : > MD5Secret : > > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : > Pickupgroup : > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > [Oct 11 18:00:48] failed to extend from 512 to 676 > === > > I'm not able to figure out what causes this and usually, I have to > prune the client, or reload sip to make it stop. > > Can someone tell me what causes it and how to fix it? > An Asterisk dynamic string (struct ast_str) needed more room for a string and failed to get it. There are many reasons why this might happen. I cannot say more without digging further into the code. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] failed to extend from 512 to 676
On Wednesday 12 October 2011 9:22:06 am Richard Mudgett wrote: > > I've got Asterisk 1.6.2.9 with RT SIP running, and for the most part, > > it's working great. > > > > However, once in a while, I'll get some strange output from "sip show > > peer .." > > > > For example: > > === > > *CLI> sip show peer 687F74D9848C-1 > > * Name : 687F74D9848C-1 > > Realtime peer: Yes, cached > > Secret : > > MD5Secret : > > > > Transfer mode: open > > CallingPres : Presentation Allowed, Not Screened > > Callgroup : > > Pickupgroup : > > [Oct 11 18:00:48] failed to extend from 512 to 676 > > [Oct 11 18:00:48] failed to extend from 512 to 676 > > [Oct 11 18:00:48] failed to extend from 512 to 676 > > [Oct 11 18:00:48] failed to extend from 512 to 676 > > [Oct 11 18:00:48] failed to extend from 512 to 676 > > [Oct 11 18:00:48] failed to extend from 512 to 676 > > [Oct 11 18:00:48] failed to extend from 512 to 676 > > === > > > > I'm not able to figure out what causes this and usually, I have to > > prune the client, or reload sip to make it stop. > > > > Can someone tell me what causes it and how to fix it? > > An Asterisk dynamic string (struct ast_str) needed more room for a string > and failed to get it. There are many reasons why this might happen. I > cannot say more without digging further into the code. > > Richard Should I worry about it? -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI not Installed?
Hello Everyone, I am trying to get AGI going. The command "agi show commands" yields: DeadCommand Description No answer Not available Yes asyncagi break Not available No channel status Not available Yes database del Not available Yes database deltree Not available Yes database get Not available Yes database put Not available Yes exec Not available No get data Not available Yes get full variable Not available No get option Not available Yes get variable Not available No hangup Not available Yes noop Not available No receive char Not available No receive text Not available Norecord file Not available No say alpha Not available No say digits Not available No say number Not available No say phonetic Not available No say date Not available No say time Not available No say datetime Not available No send image Not available No send text Not available No set autohangup Not available No set callerid Not available Noset context Not available No set extension Not available No set music Not available No set priority Not available Yes set variable Not available Nostream file Not available Nocontrol stream file Not available No tdd mode Not available Yesverbose Not available No wait for digit Not available No speech create Not available No speech set Not available Yes speech destroy Not available Nospeech load grammar Not available Yes speech unload grammar Not available Nospeech activate grammar Not available No speech deactivate grammar Not available No speech recognize Not available No gosub Not available The /var/lib/asterisk/agi-bin/ dir is empty. Is there a ./config flag needed to install AGI? Thanks in Advance, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk HoneyPot
Hi All, I'm not the first to try to start a VOIP blacklist but currently working on a project for the next 12 hours, hopefully I can get it up soon. What I intend to do is to work with a few reliable Harvester to gather the logs. A simple script to parse it then extract the list of attackers IP, compile them and send them out to the list. If any of you are kind enough to zip and send me a /var/log/asterisk/messages that contain hacker's scan & attack, it will be helpful to my research. Do email me at j...@asteriskhoneypot.com . Let me know if you are keen to be a harvester as well.Thanks. Regards, Jackster -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scheduled Maintenance for Asterisk Project community services
On Thursday, October 13th, 2011, the Asterisk community services listed below will be undergoing maintenance (software upgrades and updates). The services will be shut down at approximately 9:00 PM CDT (2:00 AM October 14 UTC), and will return no later than 10:00 PM CDT. We apologize in advance for any inconvenience this may cause. The affected services are: issues.asterisk.org/jira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Binding asterisk to two static IPs
He all, I've got a similar setup like [1], and the same issues thats described there. However, there was never a reply to this thread. I'm using a HA-cluster to run asterisk, on two servers, with two virtual ips. One for the phones to register, the other one from a different net to send and receive calls trough my provider. This is aswell a private net, without nat. >From [1]: "If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on the active node, if you have a box address of 192.168.1.101 and a floating address of 192.168.1.102, then if you use bindaddr=0.0.0.0 you will find that phones on the 192.168.1.x subnet will not register on the floating address, which of course defeats the point of HA clustering. What happens is that the registration packets go to the floating address 192.168.1.102 but the response packets appear to come from 192.168.1.101 [same NIC but the packet contains the base address attached to the NIC], so the registration fails." Any idea how to solve this? Thanks, Georg [1] http://www.fonality.com/trixbox/forums/trixbox-forums/help/binding-sip-multiple-not-all-ip-addresses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS ports on TDM410P card...
Changes so far: chan_dahdi.conf: [my-phones](!) . . . context = my-phones signalling = fxo_ks . . . [phone1](my-phones) . . . [phone2](my-phones) . . . [phone3](my-phones) . . . [phone4](my-phones) And extensions.conf is the same. Seems to be working now, good eyes. I have the O'Reilly book on Asterisk 1.8, though I'm wondering where to look in it to learn how to allow incoming calls from the phone company to ring SIP box and FXS connected handsets? This would be a neat feature, especially if there was a way from the handset to turn it off. Just to make sure I'm clear, imagine you are hooked to a Linksys PAP2 and not only want to call out via the phone company but you want to receive calls there from the phone company as well. Imagine there is an extension to call that toggles this behavior on/off. So say 2025 is the special extension which you call and a voice says relaying phone company on. You hang up then, the phone rings, and you pick up a call from somewhere remote via the phone company. You hang up when you're done and decide the behavior should be turned off, so you dial 2025 again and the voice says relaying phone company off. Now if a call is incoming from the phone company, your phone doesn't ring. You can call all local extensions and even remote numbers, but you can't receive remote calls. Another trick I want to pull is this. I have a few extensions, 2000 to 2011, where I'd like to have an extension someone can call to figure out what these extensions are. Say 1000 or even 0 if that will work. Something easy to remember anyways. Another neat trick would be to list what the extensions are when someone enters an invalid extension. Say someone dials 1011, not one of my extensions and not a remote phone number prefixed by 8 or 9. The last trick I want to pull, I want an extension that will ring inclusively 2000 to 2011, say 2012. How do I set this up by hand? Thank you again for helping me figure out the context problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
On 12.10.2011 23:27, ge...@riseup.net wrote: > "If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on > the active node, if you have a box address of 192.168.1.101 and a floating > address of 192.168.1.102, then if you use > > bindaddr=0.0.0.0 ... > Any idea how to solve this? Yes: Use "bindaddr=192.168.1.102". That's how we solved it on our Asterisk boxes. Another solution would be to use tcp, but not all SIP clients support that (and I don't know how good Asterisk does). Personally, I think this is a shortcoming in Asterisk. Every application with udp server functionality should handle this correctly. E.g. FreeRADIUS has a compile time option for this (--with-udpfromto, unfortunately off by default, for whatever reasons). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
Hallo, > On 12.10.2011 23:27, ge...@riseup.net wrote: >> "If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on >> the active node, if you have a box address of 192.168.1.101 and a >> floating >> address of 192.168.1.102, then if you use >> >> bindaddr=0.0.0.0 > ... >> Any idea how to solve this? > > Yes: Use "bindaddr=192.168.1.102". That's how we solved it on our > Asterisk boxes. Another solution would be to use tcp, but not all SIP > clients support that (and I don't know how good Asterisk does). If I use the floating internal ip, I can't reach my provider anymore. Thought this was clear. So: This is actually the problem... :) Thanks, Georg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
i had a similar challenge having Asterisk listen to multiple ports.. "some of my agents located in countries where SIP is blocked" the only effective way is to use IPTABLES i believe your problem can be solved with the same method. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 > Date: Wed, 12 Oct 2011 23:27:16 +0200 > From: ge...@riseup.net > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Binding asterisk to two static IPs > > He all, > > I've got a similar setup like [1], and the same issues thats described > there. However, there was never a reply to this thread. > > I'm using a HA-cluster to run asterisk, on two servers, with two virtual > ips. One for the phones to register, the other one from a different net to > send and receive calls trough my provider. This is aswell a private net, > without nat. > > From [1]: > > "If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on > the active node, if you have a box address of 192.168.1.101 and a floating > address of 192.168.1.102, then if you use > > bindaddr=0.0.0.0 > > you will find that phones on the 192.168.1.x subnet will not register on > the floating address, which of course defeats the point of HA clustering. > What happens is that the registration packets go to the floating address > 192.168.1.102 but the response packets appear to come from 192.168.1.101 > [same NIC but the packet contains the base address attached to the NIC], > so the registration fails." > > Any idea how to solve this? > > Thanks, > Georg > > [1] > http://www.fonality.com/trixbox/forums/trixbox-forums/help/binding-sip-multiple-not-all-ip-addresses > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
On 10/11/11 8:10 PM, Olivier wrote: > I'll start a test session in a couple of minutes and report here. > > The strangest things is this inconsistency: I can imagine million of > reasons why a number is not presented but I can't think of any > explaining why it would change in a couple of hours. Inconsistent configuration over multiple routes probably. I know I have one route (the default actually) to a number of destinations where I am 100% percent able to send redirected number information, but another route just will not pass it on to the destination. So normally calls to these destinations have nice caller id as if A was calling C (at least that's what C sees in their display) but every now and then I flow over to the alternative route and the information is lost, C doesn't see A, but B. Nothing I can do about it, been fighting over it for ages but I just doesn't seem to be able to make it work. -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI not Installed?
what version of Asterisk are you using? try issuing "agi show" from the Asterisk CLI console and see if you get some output? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 > Date: Wed, 12 Oct 2011 13:24:23 -0400 > From: sym...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] AGI not Installed? > > Hello Everyone, > > I am trying to get AGI going. The command "agi show commands" yields: > > DeadCommand Description >No answer Not available > Yes asyncagi break Not available >No channel status Not available > Yes database del Not available > Yes database deltree Not available > Yes database get Not available > Yes database put Not available > Yes exec Not available >No get data Not available > Yes get full variable Not available >No get option Not available > Yes get variable Not available >No hangup Not available > Yes noop Not available >No receive char Not available >No receive text Not available >Norecord file Not available >No say alpha Not available >No say digits Not available >No say number Not available >No say phonetic Not available >No say date Not available >No say time Not available >No say datetime Not available >No send image Not available >No send text Not available >No set autohangup Not available >No set callerid Not available >Noset context Not available >No set extension Not available >No set music Not available >No set priority Not available > Yes set variable Not available >Nostream file Not available >Nocontrol stream file Not available >No tdd mode Not available > Yesverbose Not available >No wait for digit Not available >No speech create Not available >No speech set Not available > Yes speech destroy Not available >Nospeech load grammar Not available > Yes speech unload grammar Not available >Nospeech activate grammar Not available >No speech deactivate grammar Not available >No speech recognize Not available >No gosub Not available > > > The /var/lib/asterisk/agi-bin/ dir is empty. Is there a ./config > flag needed to install AGI? > > Thanks in Advance, > > Nick. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
On 13.10.2011 00:27, ge...@riseup.net wrote: > If I use the floating internal ip, I can't reach my provider anymore. > Thought this was clear. After reading your original message, this is clear, yes. Sorry for being sloppy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
> After reading your original message, this is clear, yes. Sorry for being > sloppy. np ;) Anyone else? Would be really really great... Thanks, Georg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
On 10/12/2011 3:55 PM, ge...@riseup.net wrote: >> After reading your original message, this is clear, yes. Sorry for being >> sloppy. > > np ;) > > Anyone else? > Would be really really great... > I solved it by having two physical connections to my network. PBX E0 IP 192.168.100.36 NM 255.255.255.0 GW 192.168.100.1 E1 IP 192.168.101.254 NM 255.255.255.0 GW n/a All the phones reside withing the 192.168.101.0/24 network. I still have bindaddr=0.0.0.0 so I can talk to my provider and my phones. But on two different interfaces. That forces the communication to always come from the correct source IP addr. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ http://www.bendsource.com/ C - (541) 408-5189 O - (541) 323-9113 H - (541) 323-4219 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
> I solved it by having two physical connections to my network. Yes, I thought of this too. I used the second nic for the drbd-communication, but I think I will have to change this. Thanks, Georg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10 'database show' CLI command
Up through 1.8, 'database show' returned results ordered by key. In 10, the output is unordered (or maybe chronological?). Is this intentional? (I know 'database query' will let me view the AstDB any way I want, but the output isn't formatted as nicely.) -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] failed to extend from 512 to 676
Hi Mike, I had this problem (using realtime sip peers). I don't know if this error message (same one; failed to extend) is the cause but the site in question will build a memory leak (>3GB) and then Asterisk will become unstable (registration / qualify status would cycle quickly in large blocks) and then Asterisk would crash. On reload there is obviously no problem as the memory leak has been cleared. One of my other symptoms was that when the problem started getting worse (i.e. the memory leak was increasing) the messages would appear more and more frequently in both the log files (can't remember what the settings are) as well as the CLI if you are logged in. You woldn't need to run a command to produce the error message. My suggestion to determine if it will be a problem or not is to look at your memory usage over a period of time. I am considering upgrading the site in question to 1.8.7 to see if that fixes the problem but I have to do some more internal testing first. Kind Regards Stuart Elvish PS My issues was posted to asterisk-users under the title "Error - Failed to extend from xxx to xxx". >> > [Oct 11 18:00:48] failed to extend from 512 to 676 >> > [Oct 11 18:00:48] failed to extend from 512 to 676 >> > [Oct 11 18:00:48] failed to extend from 512 to 676 >> > [Oct 11 18:00:48] failed to extend from 512 to 676 >> > [Oct 11 18:00:48] failed to extend from 512 to 676 >> > [Oct 11 18:00:48] failed to extend from 512 to 676 >> > [Oct 11 18:00:48] failed to extend from 512 to 676 > Should I worry about it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue calls to agent end prematurely with diastatus cancel
We have a system running Asterisk-1.4.40. Queue calls are distributed using rrmemory with a 20 second timeout. What we are seeing is; when a call in the queue will call the first agent for 20 seconds, and subsequent attempts will call agents for random periods of time (as little as one second), and continue on to the same or next agent. When this happens, the dialstatus variable is set to "Cancel". This suggests that the queue is canceling the call to the agent, but we can find no configuration or error logging to show why this is happening. I also was unable to find any bugs logged on this issue. How can we further troubleshoot this issue? Chris queues.conf [myqueue] strategy = rrmemory joinempty = strict leavewhenempty = strict ringinuse=no monitor-join=yes monitor-format=wav monitor-type = MixMonitor context=ss-queueout servicelevel = 180 wrapuptime = 0 timeout = 20 retry = 0 weight = 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
